I’ve got SJPhone SIP softclients registering to Asterisk, and Asterisk registering to a Cisco VoIP GW that also acts as SIP Registrar (this topology is not for design reasons, just part of initial tinkering and learning in prep for a more rational subsequent design).
I’ve set canreinvite=yes in sip.conf for both the client side and the upstream proxy side and reloaded chan_sip.so.
ethereal trace shows RTP packets relayed via Asterisk throughout session. Looks like the sip client initiating the call sends an OPTIONS message and gets back a 404 Not Found with a list of the normal supported options.
So is it up to my client (sjphone on PC) to create a new invite to the VoIP GW? How would it know how the contact info? Same with VoIP GW… Asterisk must be the one to initiate getting itself out of the loop.
Is Asterisk supposed to send a new INVITE in both directions with updated SDP to give the client & upstream gateway each other’s contact info…
anybody with a weed wacker to help me out of the weeds would be appreciated