7 Commits (87973eecffbf400da98b67eca1dbed038af40439)

Author	SHA1	Message	Date
Kevin Harwell	a25a630659	New SIP Channel driver: Always Auth Reject ... If no matching endpoint is found for the incoming request Asterisk will respond with a 401 Unauthorized (rejecting the request), but will first challenge if no authorization creditials are given. Changes also included moving ACL options into a new global 'security' configuration section in res_sip.conf. (closes issue ASTERISK-21433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	77002bc377	Merge in current pimp_my_sip work, including: ... 1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	94ec267888	Migrate PeerStatus events to stasis, add stasis endpoints, and add chan_pjsip device state. ... (closes issue ASTERISK-21489) (closes issue ASTERISK-21503) Review: https://reviewboard.asterisk.org/r/2601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	1b5a3069f9	Add base XML documentation for res_sip ... Thanks to Brad Latus, this patch adds a significant amount much-needed documentation to res_sip. It should cover all existing configuration options currently in Asterisk trunk. Patch-by: Brad Latus (snuffy) Review: https://reviewboard.asterisk.org/r/2471/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	b90bba7a30	Stasis: Update security events to use Stasis ... Also moves ACL messages to the security topic and gets rid of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2496/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3aac3d2ee5	Tweak res_sip priority so it gets loaded first before all other SIP stuff. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	74f2318051	Merge the pimp_my_sip branch into trunk. ... The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is *not* feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago