Commit Graph

56 Commits

Author SHA1 Message Date
Rusty Newton
6b1f91b6fc 'domain_alias' config object XML help doesn't make it clear that the name used for the object is the domain alias
(issue ASTERISK-22114)
(closes issue ASTERISK-22114)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:47:34 +00:00
Rusty Newton
fdfe1ea82e xml doc changes for clarity - 'auth' config object and auth's 'auth_type' config option
(issue ASTERISK-22108)
(closes issue ASTERISK-22108)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:40:34 +00:00
Rusty Newton
115c078e9d xml doc change for transport config object - remove non-applicable warning and add text regarding Asterisk restart
(closes issue ASTERISK-22105)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:27:40 +00:00
Kinsey Moore
dfa5be76d4 Expose res_pjsip threadpool options
Expose initial size, automatic increment, maximum size, and idle
timeout as configurable parameters for the res_pjsip thread pool.

Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-06 13:08:13 +00:00
Kinsey Moore
41cd06e03f Add CLI/AMI commands to force chan_pjsip actions
For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.

This also fixes two refcounting bugs in the outbound registration code.

Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 12:40:03 +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