Hi all, We are having troubles with the 302 responses handling by asterisk 1.4. The 302 SIP responses generates an INVITE in which To and R-URI are the same, when in the RFC 3261 (8.1.3.4) RECOMMENDS reusing of the same To, From and Call-ID values than the original request. Is there any workaround to get it work in asterisk 1.4 via extensions.conf file? if not, how could it be done by hacking chan_sip.c?
Regards -- Arturo Díaz
_______________________________________________ -- 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
