I’m use asterisk 11.2.1, freepbx, linphone.
use freepbx create sip account 1000, 1001.
1000 call 1001 is ok.
1001 call 1000 is ok.
manual create sip account 1003 in sip_additinoal.conf.
execute “sip reload” in asterisk cli.
sip show peers is ok.
1003 call 1000 is ok.
1000 call 1003 , sip response is 503 service unavailable.
req:INVITE
res:401
req:ACK
req:INVITE
res:100
res:183 session progress
res:503 service unavailable
why?
Unfortunately, this forum can’t upload attachment file.