I 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 ...
Apr 25, 2019 · ccm_sip_503_service_unavailable 2801795135 0xA700003F