Hi list, does anyone of you know wether asterisk can handle SIP_INFO on pure sip calls? Is that something I have to handle in the extensions? Does asterisk hand incoming SIP_INFO over to an already connected peer? Thanks and regards,
Christophorus Laube _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
