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Age	Commit message (Collapse)	Author
2014-04-10	res_hep_pjsip: Use the channel name instead of the call ID when it is available	Matthew Jordan
	During discussions with Alexandr Dubovikov at Kamailio World, it became apparent that while the SIP call ID is a useful identifier prior to an Asterisk channel being created, it is far more preferable to use the channel name (or some channel based identifier) when the channel is available. Homer is smart enough to tie the various messages together. This patch opts to use the channel name when it is available, falling back to the call ID otherwise. ........ Merged revisions 412088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-28	res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP	Matthew Jordan
	This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a lot of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3