On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly <tammy-li...@wiztech.biz> wrote: > my bad, both sides are generating re-invites. Vitelity ignores any > inbound invites to continue call flow. to keep the call going our pbx > has to deal with their re-invites otherwise the call terminates at 30 > minutes on the dot. Our side is ignoring the inbound invites from > vitelity and that causes the call to be torn down. >
The 'directmedia' or 'canreinvite' settings only apply to Asterisk generating a re-INVITE to initiate remote packet bridging. Setting that to 'no' will only prevent Asterisk from initiating a re-INVITE to perform said bridging; it won't apply to anything else. There's a whole host of reasons why Asterisk would generate a re-INVITE. That could be due to SIP session timers, or because a change occurred in the party identification via a connected line update. Asterisk will generate re-INVITEs when that happens, and there isn't a setting that will prevent that from happening. Asterisk should have no problem accepting and handling a re-INVITE from a provider, so long as it is formed correctly. If your provider can't accept a re-INVITE being sent to them, there's something seriously wrong with that provider. This is pretty core functionality in any SIP stack. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
