On Thu, Jan 24, 2013 at 11:44 PM, Jens Jorgensen <jbj1 at ultraemail.net> wrote: > Interesting. Will pjsip end up replacing existing SIP code inside of > asterisk? Will it use the same conference bridge stuff etc? I'm curious > how it'll fit in. > Basically, Asterisk will have a new SIP channel driver[1] based on pjsip. Digium is driving development of it. As for me, I'm just helping setup git repos, so the process to merge any patches into upstream is easier. In fact, it would be awesome if we could create an official git repo mirror on github.com. I don't have a problem doing the work, just want to make sure the location of the repo is good for everybody. [1] https://wiki.asterisk.org/wiki/display/AST/New+SIP+Channel+Driver+Architecture -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
