Commit Graph

5 Commits

Author SHA1 Message Date
Kevin Harwell 9bad1dabcf Add a reloadable option for sorcery type objects
Some configuration objects currently won't place nice if reloaded.
Specifically, in this case the pjsip transport objects.  Now when
registering an object in sorcery one may specify that the object is
allowed to be reloaded or not.  If the object is set to not reload
then upon reloading of the configuration the objects of that type
will not be reloaded.  The initially loaded objects of that type
however will remain.

While the transport objects will not longer be reloaded it is still
possible for a user to configure an endpoint to an invalid transport.
A couple of log messages were added to help diagnose this problem if
it occurs.

(closes issue ASTERISK-22382)
Reported by: Rusty Newton
(closes issue ASTERISK-22384)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2807/
........

Merged revisions 398139 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30 19:55:56 +00:00
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
2013-08-06 12:39:27 +00:00
Matthew Jordan 80c9ad102e Add AMI registration events for PJSIP outbound registration attempts
This patch adds AMI events whenever an outbound registration attempt succeeds
or fails from res_pjsip_outbound_registration. This brings it inline with
the existing SIP channel driver and IAX channel driver.

Review: https://reviewboard.asterisk.org/r/2729/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 19:01:45 +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