hello, i am tom, and new to this voip,
i have installed asterisk in aws, using freepbx to configure.
i make 2 extension chan sip in free pbx
extension : 111
password : 111
and the second is 222
in sipdroid
authorization username :111
password : 111
server : my asw ip server,port default port in freepbx sip setting : 5061
and sipdroid is registered with green light, but when i try to call from 111 to 222 i got this error in asterisk console :
[2017-12-22 02:56:47] WARNING[22013][C-00000007]: ccss.c:1011 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2017-12-22 02:56:47] WARNING[22013][C-00000007]: ccss.c:1011 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2017-12-22 02:56:47] WARNING[22013][C-00000007]: func_presencestate.c:131 presence_read: PRESENCE_STATE unknown
[2017-12-22 02:56:47] WARNING[22013][C-00000007]: app_dial.c:2421 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[2017-12-22 02:56:53] WARNING[21073]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission 343775439758@ 192.168.1.17 for seqno 2 (Critical Response)
any one pls help me, what can i do to fix this