res_hep_pjsip: Use the channel name instead of the call ID when it is available

res_hep_pjsip: Use the channel name instead of the call ID when it is available

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@412088 65c4cc65-6c06-0410-ace0-fbb531ad65f3