On 14-03-03 12:51 PM, André Valentin wrote: > Hi! Hola,
> 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? What's the exact configuration in use? Do you have a transport explicitly specified for the endpoint? Doing so will currently cause it to try to create a new connection [1]. [1] https://issues.asterisk.org/jira/browse/ASTERISK-22658 Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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