asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	8049bf94f7	Handle default body types for SIP event packages in res_pjsip_pubsub Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	5b3441ae55	Fix crash in res_pjsip_outbound_registration when the remote server can not be resolved. This crash was caused by decrementing the reference count of a newly created message when it should not be. This change fixes that but also fixes all other cases where this was incorrectly done. (closes issue ASTERISK-22188) Reported by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 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

8049bf94f7

Handle default body types for SIP event packages in res_pjsip_pubsub

Prior to this change, we would reject SUBSCRIBE requests that had no Accept
headers. Now event package handlers that handle the default type for the
event package indicate that they do so. Therefore, if we have a handler that
can handle the default type, we can allow SUBSCRIBEs for the handler's event
package that have no Accept headers.

(closes issue ASTERISK-22067)
reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/2774


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Joshua Colp

5b3441ae55

Fix crash in res_pjsip_outbound_registration when the remote server can not be resolved.

This crash was caused by decrementing the reference count of a newly created message when
it should not be. This change fixes that but also fixes all other cases where this was
incorrectly done.

(closes issue ASTERISK-22188)
Reported by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 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 (4ca0f222e85554e24d5cfb7d48b3a774d96a6e45)