On Wed, 2008-09-10 at 17:22 +0000, Scott Lawrence wrote: > So... if you include a User address in a paging group, and you put that > address on multiple phones, then they will all get the call and the > first one to auto-answer wins: that call will get the audio stream and > the others will not.
I realized that I forgot to address the other possible approach Mark raised in XECS-1652: > > Technically speaking there is a way to solve this problem with SIP. It > would be using the same mechanism RLS server uses to create dialog > subscrptions with individual phones: > > Create a reg event subscription with registrar against each user in > the paging zone. This allows paging server to track all phone > registrations for all users in the paging zone. Upon incoming paging > call create separate INVITE dialogs towards each registered phone. > This way no paralell forking is done and calls to multiple phones are > made at the same time. I'm not sure this approach will work (for either the paging server or RLS, actually) if the phone is remote. The problem is that the registration event will give you the Contact for the phone, but not the associated route set. If the phone is remote, and needs to go through a particular proxy, then using only the Contact address will sometimes not work. Unless the Contact value is a gruu (is globally routable), using it as the target for an out-of-dialog request is not a reliable thing to do. _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
