In Asterisk, sip.conf peer matching is done by source host and port unless the 'type' is set to 'dynamic'. That said, if your desire is to change the From header display name or user part of the From URI, you should use the 'uac' module and look into the 'uac_replace_from()' function.  It's a little unorthodox by RFC 2543 standards, but because it spoofs the From header statefully and reverts the value on replies before passing back to the near end--such that the initiating endpoint is none the wiser--it works. Otherwise, if your main desire is to customise entry points into various dial plan contexts for otherwise similar calls, I suggest routing on a different criterion.  Custom headers have sort of become a de facto means of out-of-band inter-machine associated signaling additions in SIP where it is possible to add them.  Why not use a custom header?  In Kamailio, add a header to the initial INVITE request that is something like:     append_hf("X-JR-Dialplan-Target: my_context"); Then, route all calls in Asterisk via one sip.conf peer and into one "master" dial plan context, where you can then make further routing decisions based on the value of the header - sip.conf:     [my_proxy]     ...     host=xx.xx.xx.xx     insecure=port,invite     ...     context=master_dp_router And then in extensions.conf:     [master_dp_router]     exten => _.,1,Goto(${SIP_HEADER(X-JR-Dialplan-Target)},${EXTEN},1)     ...     [my_context]     exten => 5551212,1,Answer     exten => 5551212,n,Playback(hello-world)     ... Alex Balashov - Principal Evariste Systems LLC