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 D?az -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090223/2733ee00/attachment.htm