On Mon, 2006-07-31 at 11:10 +0100, Alistair Cunningham wrote: > We have a customer who would like to do RTP directly between SIP > devices. The devices are not registered directly to Asterisk, but to SER > on another machine. > > It seems in this case "canreinvite = yes" is never used. Does anyone > know of a way of persuading Asterisk to issue re-invites in this case?
Although not clear from your posting I assume that the call between the two phones is setup through the Asterisk server. Asterisk will not let go if you have ie the "T" or "t" option in your Dial statement. Remove those for starters. If Asterisk is not involved at all I guess you need to find out what the equivalent of "canreinvite=yes" is in SER country. Regards, Patrick _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
