On Tue, 2021-12-21 at 09:45 -0400, Joshua C. Colp wrote: > No. Session timers on the endpoint is the closest thing to making > sure a call is active and keeping things open but does not use > OPTIONS. Note that if you're sending calls to them, then without > OPTIONS outside of calls any NAT mapping would go away unless they > re-register frequently. If they did re-register frequently then you > likely wouldn't need either.
Hi, the example I'm testing with is with sending a call to Twilio. SIP timers look perfect for this, except that after the first refresh, Twilio turns them off :( What I'm seeing is after a minute we send a re-invite with these headers: Session-Expires: 120;refresher=uac. Min-SE: 90. and the 200 OK coming back from Twilio omits them. Asterisk then doesn't send any more. This is not very helpful of them. If the callee hangs up after a while, our system doesn't notice because our firewall blocks the BYE. We can't leave these servers open to the world so need somehow to find a way of keeping the firewall open for any calls we send out. Any idea how we might solve that? -- Cheers, Kingsley. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
