Hi! I'm just trying to move my function ality from chan_sip to pjsip. I stumbled upon one problem. With chan_sip and a via persistant TLs connected phone everything works as expected. Calls in/out work. Even if asterisk tries to reach the phone, it reuses the existing TLS connection.
If I switch this to PJSIP, it stops working. I configured the following parameters: symmetric_rtp=true force_rport=true and others... I I know call the phone via PJSIP, asterisk does not reuse the TLS connection. It tries to create a new one, which of course fails. Any ideas? With kind regards, André -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
