[EMAIL PROTECTED] wrote: > Actually what happens is that from SER debug I can see the call is > looping between Asterisk and SER. but adding a number makes no > loops.
Check what the origin (IP/DNS name) of the incoming SIP message is. If it's from asterisk, send it to the user, if it is not from asterisk, it must be meant to go to asterisk. Add a couple of other tests (known user, etc) to it and then I think you'll have what you're looking for. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
