Asterisk kamailio


The Contact header in an outgoing request is a SIP URI that Asterisk can be reached at for subsequent in-dialog requests. It should not be the endpoint that is being called.

The rewrite_contact option rewrites incoming Contact headers to the actual source IP address and port which helps with NAT.


Found out there was some issue with my kamailio configuration which was causing the loop. Now everything is working fine.



working fine with PJSIP_DIAL_CONTACTS o using DIAL?



I am using Dial function in extension.conf file