I’m triying to make call from my telephone(SIP/3020) to my softphone(SIP/3028) using ATA Cisco SPA 8000 is ok, the call is made.
But the problem is, if I make call from my softphone(SIP/3028) to my telephone(SIP/3020) is not possible.
Please, see by CLI
== Using SIP RTP CoS mark 5 -- Executing [3028@ramais:1] Dial("SIP/3020-00000013", "SIP/3028,60,tT") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/3028 -- SIP/3028-00000014 is ringing -- Got SIP response 486 "Busy Here" back from XXX.X.XXX.X:36980 -- SIP/3028-00000014 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/3020-00000013' status is 'BUSY' == Using SIP RTP CoS mark 5 -- Executing [3020@ramais:1] Dial("SIP/3028-00000015", "SIP/3020,60,tT") in new stack [Apr 30 16:39:10] WARNING[C-0000000f]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/3028-00000015' status is 'CHANUNAVAIL'
My Asterisk server is in one network, and my ATA and softphone in other.
Is a problem in my NAT configuration?
Thanks in advanced!